[linux-audio-user] RAW to [other format] audio coverter questions

Rick Wright riwright at vt.edu
Wed Nov 22 11:59:48 EST 2006


Hi List,

I'm currently trying to convert raw audio data from 64bit float (ieee 
double precision) Big Endian to other formats (i.e. WAV, OGG, etc.) for 
output using some linux audio player.  The problem I'm running into is 
that when importing this data into various programs (see below), the 
audio clips outside of +/- magnitude=1.  I don't understand why this is 
and, more importantly, how to avoid/workaround this limitation.

So far I've tried and been able to successfully import the data in SoX, 
Audacity, Rezound.  For each of these programs, I've simply had to 
specify RAW format, 64bit float, sample rate, # channels, Big Endian.  
Using sox, I've successfully resampled, written other RAW files, written 
WAV files....no problem.  When the original raw audio file contains data 
that is constrained within +/- 1 magnitude scale (i.e. 0.5 peak 
magnitude sine wave w/no DC offset), there is no problem.

Questions:
1) is there something fundamentally incompatible with my original data 
and "standard" audio data wrt range of values?
2) is the reason I'm seeing all of these programs clip my data at +/- 1 
amplitude because they all use libraries with this limitation?


What I really want is a simple way to convert data from my original 
format to other "playable" formats.  SoX seems perfect if not for the 
clipping issue.  Though, I'd appreciate advice on other suitable 
applications.

Any/All help greatly appreciated!

Rick



More information about the Linux-audio-user mailing list